Asterisk via MRCP Media Relay Control Protocol

UniMRCP for Asterisk Installation

We well use Asterisk 18 in our example. Asterisk 19 is not supported anymore and Asterisk 20 was not tested with UniMRCP.So the instructions to install are below:

Note: UniMRCP server should already be installed and working.

bash

apt -y install build-essential git curl wget libnewt-dev libssl-dev libncurses5-dev subversion libsqlite3-dev libjansson-dev libxml2-dev uuid-dev default-libmysqlclient-dev
groupadd asterisk
useradd -r -d /var/lib/asterisk -g asterisk asterisk
wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18-current.tar.gz
tar -zxvf asterisk-18-current.tar.gz
cd asterisk-18.23.1
./contrib/scripts/get_mp3_source.sh
./contrib/scripts/install_prereq install
./configure --with-jansson-bundled --with-pjproject-bundled
make
make install
make samples
make config
make install-headers
ldconfig
chown -R asterisk.asterisk /etc/asterisk
chown -R asterisk.asterisk /var/{lib,log,spool}/asterisk
chown -R asterisk.asterisk /usr/lib/asterisk
systemctl enable asterisk
systemctl start asterisk

Asterisk UniMRCP installation

curl -O -J -L https://www.unimrcp.org/project/component-view/asterisk-unimrcp-1-10-0-tar-gz/download
./bootstrap
./configure
make
make install

Asterisk UniMRCP Configuration

There are two ways to use Asterisk with UniMRCP. The old way using the module res_speech_unimrcp or the new way using app_unimrcp. The latest is a bit more flexible. The configuration diagram is below.

Please change the environment and IP address according to your own installation.

Generic Speech Recognition API

This method uses res_speech_unimrcp module. In this cases you have to configure the unimrcpclient.xml and the file res_speech_unimrcp.conf..

/usr/local/unimrcp/conf/unimrcpclient.xml

xml

<?xml version="1.0" encoding="UTF-8"?>
<!-- UniMRCP client document -->
<unimrcpclient xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
               xsi:noNamespaceSchemaLocation="unimrcpclient.xsd"
               version="1.0"
               subfolder="client-profiles">
  <properties>
    <ip>192.168.86.39</ip>
    <server-ip>192.168.86.39</server-ip>
  </properties>
  <components>
    <!-- Factory of MRCP resources -->
    <resource-factory>
      <resource id="speechsynth" enable="true"/>
      <resource id="speechrecog" enable="true"/>
      <resource id="recorder" enable="true"/>
      <resource id="speakverify" enable="true"/>
    </resource-factory>

    <sip-uac id="SIP-Agent-1" type="SofiaSIP">
      <sip-ip>192.168.86.39</sip-ip>
      <sip-port>8062</sip-port>
      <sip-transport>udp</sip-transport>
      <ua-name>UniMRCP SofiaSIP</ua-name>
      <sdp-origin>UniMRCPClient</sdp-origin>
    </sip-uac>

    <!-- UniRTSP MRCPv1 signaling agent -->
    <rtsp-uac id="RTSP-Agent-1" type="UniRTSP">
      <max-connection-count>100</max-connection-count>
      <sdp-origin>UniMRCPClient</sdp-origin>
    </rtsp-uac>

    <!-- MRCPv2 connection agent -->
    <mrcpv2-uac id="MRCPv2-Agent-1">
      <max-connection-count>100</max-connection-count>
      <max-shared-use-count>100</max-shared-use-count>
      <offer-new-connection>false</offer-new-connection>
      <rx-buffer-size>1024</rx-buffer-size>
      <tx-buffer-size>1024</tx-buffer-size>
    </mrcpv2-uac>

    <!-- Media processing engine -->
    <media-engine id="Media-Engine-1">
      <realtime-rate>1</realtime-rate>
    </media-engine>

    <!-- Factory of RTP terminations -->
    <rtp-factory id="RTP-Factory-1">
      <rtp-port-min>4000</rtp-port-min>
      <rtp-port-max>5000</rtp-port-max>
    </rtp-factory>
  </components>

  <settings>
    <!-- Common (default) RTP/RTCP settings -->
    <rtp-settings id="RTP-Settings-1">
      <jitter-buffer>
        <adaptive>1</adaptive>
        <playout-delay>50</playout-delay>
        <max-playout-delay>600</max-playout-delay>
        <time-skew-detection>1</time-skew-detection>
      </jitter-buffer>
      <ptime>20</ptime>
      <codecs>PCMU PCMA L16/96/8000 PCMU/97/16000 PCMA/98/16000 L16/99/16000</codecs>

      <!-- Enable/disable RTCP support -->
      <rtcp enable="false">
        <rtcp-bye>1</rtcp-bye>
        <tx-interval>5000</tx-interval>
        <rx-resolution>1000</rx-resolution>
      </rtcp>
    </rtp-settings>
  </settings>
</unimrcpclient>

/etc/asterisk/res-speech-unimrcp.conf

ini

[general]
unimrcp-profile = uni2      ; UniMRCP MRCPv2 Server
log-level = DEBUG

[grammars]
[mrcpv2-properties]
Recognition-Timeout = 20000
No-Input-Timeout = 15000

[mrcpv1-properties]
Recognition-Timeout = 20000
No-Input-Timeout = 15000

MRCP Applications

Você deve escolher uma ou outra forma de trabalhar. Com o sistema genérico ou com as aplicações de UniMRCP. Você pode trabalhar com os dois ao mesmo tempo, mas vai ter de configurar ambos. A aplicação MRCP é mais simples de configurar, bast configurar o arquivo mrcp.conf.

ini

[general]
; Default ASR and TTS profiles.
default-asr-profile = uni2
default-tts-profile = uni2
log-level = DEBUG
max-connection-count = 100
max-shared-count = 100
offer-new-connection = 1

[uni2]
; MRCP settings
version = 2
;
; SIP settings
server-ip = 192.168.86.39
server-port = 8060
; SIP user agent
client-ip = 192.168.86.39
client-port = 8063
sip-transport = udp
;
; RTP factory
rtp-ip = 192.168.86.39
rtp-port-min = 28000
rtp-port-max = 29000
;
; Jitter buffer settings
playout-delay = 50
max-playout-delay = 200
; RTP settings
ptime = 20
codecs=G722 PCMU telephone-event/101/8000
; RTCP settings
rtcp = 0

Testing the Speech Recognition

Once you have installed and configure Asterisk res-speech-unimrcp.conf, please restart your asterisk server. You can find a lot more examples at:

UniMRCP for Asterisk Manual

ini

[default]
exten => 700,1,goto(res-speech-unimrcp-1)
exten => 701,1,goto(synth-unimrcp-1)
exten => 702,1,goto(synthandrecog-unimrcp-1)

[res-speech-unimrcp-1]
exten => s,1,Answer()
exten => s,2,SpeechCreate()
exten => s,3,NoOP()
exten => s,4,SpeechActivateGrammar(builtin:speech/transcribe?language=en)
exten => s,5,SpeechBackground(beep, 20)

exten => s,6,Verbose(1, "Recognition result count: ${SPEECH(results)}")
exten => s,7,GotoIf($["${SPEECH(results)}" = "0"]?8:10)
exten => s,8,Verbose(1, "Failed to recognize")
exten => s,9,Goto(30)
exten => s,10,Verbose(1, "Recognition result: ${SPEECH_TEXT(0)}, confidence score: ${SPEECH_SCORE(0)}, grammar-uri: ${SPEECH_GRAMMAR(0)}")
exten => s,11,SpeechDestroy()
exten => s,12,Hangup()

[res-synth-unimrcp-1]
exten => s,1,Answer
exten => s,n,MRCPSynth(I am unimrcp, the most mature way to synthsize voice!,v=en-US-JennyNeural)

[synth-unimrcp-1]
exten => s,1,Answer
exten => s,n,MRCPSynth(I am unimrcp, the most mature way to synthsize voice!,v=en-US-JennyNeural)

[synthandrecog-unimrcp-1]
exten => s,1,Answer
exten => s,n,SynthAndRecog(Please say a number,builtin:grammar/number,t=5000&b=1&ct=0.7&spl=en-US)
exten => s,n,Verbose(1, ${RECOG_STATUS}, ${RECOG_COMPLETION_CAUSE}, ${RECOG_RESULT})
exten => s,n,Hangup

Asterisk via MRCP Media Relay Control Protocol ​

UniMRCP for Asterisk Installation ​

Asterisk UniMRCP installation ​

Asterisk UniMRCP Configuration ​

Generic Speech Recognition API ​

MRCP Applications ​

Testing the Speech Recognition ​